One document matched: draft-ietf-rtcweb-security-arch-06.xml
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<rfc category="std" docName="draft-ietf-rtcweb-security-arch-06"
ipr="pre5378Trust200902">
<front>
<title abbrev="RTCWEB Sec. Arch.">RTCWEB Security Architecture</title>
<author fullname="Eric Rescorla" initials="E.K." surname="Rescorla">
<organization>RTFM, Inc.</organization>
<address>
<postal>
<street>2064 Edgewood Drive</street>
<city>Palo Alto</city>
<region>CA</region>
<code>94303</code>
<country>USA</country>
</postal>
<phone>+1 650 678 2350</phone>
<email>ekr@rtfm.com</email>
</address>
</author>
<date day="22" month="January" year="2013" />
<area>RAI</area>
<workgroup>RTCWEB</workgroup>
<abstract>
<t>
The Real-Time Communications on the Web (RTCWEB) working group
is tasked with standardizing protocols for enabling real-time
communications within user-agents using web technologies (e.g
JavaScript). The
major use cases for RTCWEB technology are real-time audio and/or video calls,
Web conferencing, and direct data transfer. Unlike most conventional real-time systems
(e.g., SIP-based soft phones) RTCWEB communications are directly controlled
by some Web server, which poses new security challenges.
For instance, a Web browser might expose a JavaScript
API which allows a server to place a video call. Unrestricted access to such
an API would allow any site which a user visited to "bug" a user's computer,
capturing any activity which passed in front of their camera.
[I-D.ietf-rtcweb-security] defines the RTCWEB
threat model. This document
defines an architecture which provides security within that threat model.
</t>
</abstract>
<note title="Legal">
<t>THIS DOCUMENT AND THE INFORMATION CONTAINED THEREIN ARE PROVIDED ON
AN “AS IS” BASIS AND THE CONTRIBUTOR, THE ORGANIZATION
HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE
IETF TRUST, AND THE INTERNET ENGINEERING TASK FORCE, DISCLAIM ALL
WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY
WARRANTY THAT THE USE OF THE INFORMATION THEREIN WILL NOT INFRINGE ANY
RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A
PARTICULAR PURPOSE.</t>
</note>
</front>
<middle>
<section title="Introduction" anchor="sec.introduction">
<t>
The Real-Time Communications on the Web (RTCWEB) working
group is tasked with standardizing protocols for real-time
communications between Web browsers. The major use cases for RTCWEB
technology are real-time audio and/or video calls, Web conferencing,
and direct data transfer. Unlike most conventional real-time systems,
(e.g., SIP-based<xref target="RFC3261"></xref> soft phones) RTCWEB
communications are directly controlled by some Web server, as shown in
<xref target="fig.simple"/>.
</t>
<figure title="A simple RTCWEB system" anchor="fig.simple">
<artwork><![CDATA[
+----------------+
| |
| Web Server |
| |
+----------------+
^ ^
/ \
HTTP / \ HTTP
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------->| Browser |
| | | |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
A more complicated system might allow for interdomain calling,
as shown in <xref target="fig.multidomain"/>.
The protocol to be used between the domains
is not standardized by RTCWEB, but given the installed
base and the form of the RTCWEB API is likely to be something
SDP-based like SIP or XMPP.
</t>
<figure title="A multidomain RTCWEB system" anchor="fig.multidomain">
<artwork><![CDATA[
+--------------+ +--------------+
| | SIP,XMPP,...| |
| Web Server |<----------->| Web Server |
| | | |
+--------------+ +--------------+
^ ^
| |
HTTP | | HTTP
| |
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
| Browser |<---------------->| Browser |
| | | |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
This system presents a number of new security challenges,
which are analyzed in <xref target="I-D.ietf-rtcweb-security"/>.
This document describes a security architecture for RTCWEB
which addresses the threats and requirements described in
that document.
</t>
</section>
<section anchor="sec-term" title="Terminology">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
<section title="Trust Model" anchor="sec.proposal.trusthierarchy">
<t>
The basic assumption of this architecture is that network resources
exist in a hierarchy of trust, rooted in the browser, which
serves as the user's TRUSTED COMPUTING BASE (TCB). Any security
property which the user wishes to have enforced must be
ultimately guaranteed by the browser (or transitively by
some property the browser verifies). Conversely, if the
browser is compromised, then no security guarantees are possible.
Note that there are cases (e.g., Internet kiosks) where the
user can't really trust the browser that much. In these cases,
the level of security provided is limited by how much they
trust the browser.
</t>
<t>
Optimally, we would not rely on trust in any entities other
than the browser. However, this is unfortunately not possible
if we wish to have a functional system.
Other network elements fall into two categories: those which
can be authenticated by the browser and thus are partly trusted--though
to the minimum extent necessary--and
those which cannot be authenticated and thus are untrusted.
This is a natural extension of the end-to-end principle.
</t>
<section title="Authenticated Entities" anchor="sec.proposal.authenticated">
<t>
There are two major classes of authenticated entities in the system:
</t>
<t>
<list style="symbols">
<t>Calling services: Web sites whose origin we can verify
(optimally via HTTPS, but in some cases because we are on
a topologically restricted network, such as behind a firewall).</t>
<t>Other users: RTCWEB peers whose origin we can verify
cryptographically (optimally via DTLS-SRTP).</t>
</list>
</t>
<t>
Note that merely being authenticated does not make these
entities trusted. For instance, just because we can verify
that https://www.evil.org/ is owned by Dr. Evil does not
mean that we can trust Dr. Evil to access our camera
and microphone. However, it gives the user an opportunity
to determine whether he wishes to trust Dr. Evil or not;
after all, if he desires to contact Dr. Evil (perhaps
to arrange for ransom payment), it's safe
to temporarily give him access to the camera and microphone
for the purpose of the call, but he doesn't want
Dr. Evil to be able to access his camera and
microphone other than during the call. The point here is that we must
first identify other elements before we can determine whether
and how much to trust them.
</t>
<t>
It's also worth noting that there are settings where
authentication is non-cryptographic, such as other machines
behind a firewall. Naturally, the level of trust one can
have in identities verified in this way depends on how
strong the topology enforcement is.
</t>
</section>
<section title="Unauthenticated Entities" anchor="sec.proposal.unauthenticated">
<t>
Other than the above entities, we are not generally able to
identify other network elements, thus we cannot trust them.
This does not mean that it is not possible to have any interaction
with them, but it means that we must assume that they will
behave maliciously and design a system which is secure even
if they do so.
</t>
</section>
</section>
<!-- Not layered ? -->
<section title="Overview" anchor="sec.proposal.overview">
<!-- TODO: Federated -->
<t>
This section describes a typical RTCWeb session and shows how
the various security elements interact and what guarantees are
provided to the user. The example in this section is a "best case"
scenario in which we provide the maximal amount of user
authentication and media privacy with the minimal level of trust in
the calling service. Simpler versions with lower levels of
security are also possible and are noted in the text where
applicable. It's also important to recognize the tension
between security (or performance) and privacy. The example
shown here is aimed towards settings where we are more concerned
about secure calling than about privacy, but as we shall
see, there are settings where one might wish to make different
tradeoffs--this architecture is still compatible with those
settings.
</t>
<t>
For the purposes of this example, we assume the topology shown
in the figures below. This topology is derived from the
topology shown in <xref target="fig.simple"/>, but separates
Alice and Bob's identities from the process of signaling.
Specifically, Alice and Bob have relationships with
some Identity Provider (IdP) that supports a protocol such
as OpenID or BrowserID) that can be used to demonstrate
their identity to other parties. For instance, Alice might
have an account with a social network which she can then
use to authenticate to other web sites without explicitly
having an account with those sites; this is a fairly conventional
pattern on the Web. <xref target="sec.trust-relationships"/>
provides an overview of Identity Providers and the relevant terminology.
</t>
<t>
This separation of identity provision and signaling
isn't particularly important in "closed world"
cases where Alice and Bob are users on the same social network and have
identities based on that domain (<xref target="fig.proposal.idp"/>)
However, there are important
settings where that is not the case, such as
federation (calls from one domain to another; <xref target="fig.proposal-federated.idp"/>) and
calling on untrusted sites, such as where two users who have
a relationship via a given social network want to call each
other on another, untrusted, site, such as a poker site.
</t>
<t>
Note that the servers themselves are also authenticated by an external
identity service, the SSL/TLS certificate infrastructure (not shown).
As is conventional in the Web, all identities are ultimately rooted
that system. For instance, when an IdP makes an identity assertion,
the Relying Party consuming that assertion is able to verify because
it is able to connect to the IdP via HTTPS.
</t>
<figure title="A call with IdP-based identity" anchor="fig.proposal.idp">
<artwork><![CDATA[
+----------------+
| |
| Signaling |
| Server |
| |
+----------------+
^ ^
/ \
HTTPS / \ HTTPS
/ \
/ \
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<---------->| Browser | Bob
| | (DTLS-SRTP)| |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<--------+ | |
| IdP | | | IdP |
| | +------->| |
+-----------+ +-----------+
]]></artwork>
</figure>
<t>
<xref target="fig.proposal-federated.idp"/> shows essentially
the same calling scenario but with a call between two
separate domains (i.e., a federated case). As mentioned
above, the domains communicate by some unspecified
protocol and providing separate signaling and identity
allows for calls to be authenticated regardless of
the details of the inter-domain protocol.
</t>
<figure title="A federated call with IdP-based identity" anchor="fig.proposal-federated.idp">
<artwork><![CDATA[
+----------------+ Unspecified +----------------+
| | protocol | |
| Signaling |<----------------->| Signaling |
| Server | (SIP, XMPP, ...) | Server |
| | | |
+----------------+ +----------------+
^ ^
| |
HTTPS | | HTTPS
| |
| |
v v
JS API JS API
+-----------+ +-----------+
| | Media | |
Alice | Browser |<--------------------------->| Browser | Bob
| | DTLS-SRTP | |
+-----------+ +-----------+
^ ^--+ +--^ ^
| | | |
v | | v
+-----------+ | | +-----------+
| |<-------------------------+ | |
| IdP | | | IdP |
| | +------------------------>| |
+-----------+ +-----------+
]]></artwork>
</figure>
<section title="Initial Signaling">
<t>
For simplicity, assume the topology in <xref target="fig.proposal.idp"/>.
Alice and Bob are both users of a common calling service; they
both have approved the calling service to make calls (we
defer the discussion of device access permissions till later).
They are both connected to the calling service via HTTPS
and so know the origin with some level of confidence. They also
have accounts with some identity provider.
This sort of identity service is becoming increasingly
common in the Web environment in technologies such
(BrowserID, Federated Google Login,
Facebook Connect, OAuth, OpenID, WebFinger), and
is often provided as a side effect service of a user's ordinary
accounts with some service. In this example, we show Alice and
Bob using a separate identity service, though the identity service
may be the same entity as the calling service or there may
be no identity service at all.
</t>
<t>
Alice is logged onto the calling service and decides to call Bob.
She can see from the calling service that he is online and the
calling service presents a JS UI in the form of a button
next to Bob's name which says "Call". Alice clicks the button,
which initiates a JS callback that instantiates a PeerConnection
object. This does not require a security check: JS from any
origin is allowed to get this far.
</t>
<t>
Once the PeerConnection is created, the calling service JS
needs to set up some media. Because this is an audio/video
call, it creates two MediaStreams, one connected to an
audio input and one connected to a video input. At this
point the first security check is required: untrusted
origins are not allowed to access the camera and microphone.
In this case, because Alice is a long-term user of the
calling service, she has made a permissions grant (i.e.,
a setting in the browser) to
allow the calling service to access her camera and microphone
any time it wants. The browser checks this setting when the
camera and microphone requests are made and thus allows them.
</t>
<t>
In the current W3C API, once some streams have been added,
Alice's browser + JS generates a signaling message
<xref target="I-D.ietf-rtcweb-jsep"/> containing:
</t>
<t>
<list style="symbols">
<t>Media channel information</t>
<t>ICE candidates</t>
<t>A fingerprint attribute binding the communication to a key pair
<xref target="RFC5763"/>. Note that this key may simply be
ephemerally generated for this call or specific to this domain,
and Alice may have a large number of such keys.</t>
</list>
</t>
<t>
Prior to sending out the signaling message, the PeerConnection code
contacts the identity service and obtains an assertion binding
Alice's identity to her fingerprint. The exact details depend on
the identity service (though as discussed in <xref target="sec.generic.idp"/>
PeerConnection can be agnostic to them), but for now it's
easiest to think of as a BrowserID assertion.
The assertion may bind other information to the identity besides
the fingerprint, but at minimum it needs to bind
the fingerprint.
</t>
<t>
This message is sent to the signaling server, e.g., by XMLHttpRequest
<xref target="XmlHttpRequest"/> or by WebSockets <xref target="RFC6455"/>
The signaling server processes the message from Alice's browser,
determines that this is a call to Bob and sends a signaling
message to Bob's browser (again, the format is currently undefined).
The JS on Bob's browser processes it, and alerts Bob to the incoming
call and to Alice's identity. In this case, Alice has provided an
identity assertion and so Bob's browser contacts Alice's identity provider
(again, this is done in a generic way so the browser has no
specific knowledge of the IdP) to verify the assertion. This
allows the browser to display a trusted element in the browser chrome
indicating that a call is coming in from Alice. If Alice is in Bob's address book,
then this interface might also include her real name, a picture, etc.
The calling site will also provide
some user interface element (e.g., a button) to allow Bob to
answer the call, though this is most likely not part of the
trusted UI.
</t>
<t>
If Bob agrees [I am ignoring early media for now],
a PeerConnection is instantiated with the message from Alice's side.
Then, a similar process
occurs as on Alice's browser: Bob's browser verifies that the calling
service is approved, the media streams are created, and a return
signaling message containing media information, ICE candidates, and
a fingerprint is sent back to Alice via the signaling service.
If Bob has a relationship with an IdP, the message will also come
with an identity assertion.
</t>
<t>
At this point, Alice and Bob each know that the other party wants to
have a secure call with them. Based purely on the interface provided
by the signaling server, they know that the signaling server claims
that the call is from Alice to Bob. This level of security is provided
merely by having the fingerprint in the message and having that message
received securely from the signaling server.
Because the far end sent an identity
assertion along with their message, they know that this is verifiable
from the IdP as well. Note that if the call is federated, as shown in
<xref target="fig.proposal-federated.idp"/> then Alice is able to
verify Bob's identity in a way that is not mediated by either
her signaling server or Bob's. Rather, she verifies it directly
with Bob's IdP.
</t>
<t>
Of course, the call works perfectly well if
either Alice or Bob doesn't have a relationship with an IdP; they
just get a lower level of assurance. I.e., they simply
have whatever information their calling site claims about
the caller/calllee's identity.
Moreover, Alice might wish
to make an anonymous call through an anonymous calling site,
in which case she would of course just not provide any identity
assertion and the calling site would mask her identity from Bob.
</t>
</section>
<section title="Media Consent Verification">
<t>
As described in
(<xref target="I-D.ietf-rtcweb-security"/>; Section 4.2)
media consent verification is provided via ICE.
Thus, Alice and Bob perform ICE checks with each other.
At the completion of these checks, they are ready to
send non-ICE data.
</t>
<t>
At this point, Alice knows that (a) Bob (assuming he is verified
via his IdP) or someone else who the
signaling service is claiming is Bob is willing to exchange
traffic with her and (b) that either Bob is at the IP address
which she has verified via ICE or there is an attacker who
is on-path to that IP address detouring the traffic. Note that
it is not possible for an attacker who is on-path between Alice
and Bob but not
attached to the signaling service to spoof these checks
because they do not have the ICE credentials. Bob's has the
same security guarantees with respect to Alice.
</t>
</section>
<section title="DTLS Handshake">
<t>
Once the ICE checks have completed [more specifically, once some
ICE checks have completed], Alice and Bob can set up a secure
channel. This is performed via DTLS <xref target="RFC4347"/>
(for the data channel) and DTLS-SRTP <xref target="RFC5763"/>
for the media channel. Specifically, Alice and Bob perform
a DTLS handshake on every channel which has been established
by ICE. The total number of channels depends on the amount of muxing;
in the most likely case we are using both RTP/RTCP mux and
muxing multiple media streams on the same channel, in which
case there is only one DTLS handshake. Once the DTLS handshake
has completed, the keys are exported
<xref target="RFC5705"/> and used to key SRTP
for the media channels.
</t>
<t>
At this point, Alice and Bob know that they share a set
of secure data and/or media channels with keys which are
not known to any third-party attacker. If Alice and
Bob authenticated via their IdPs, then they also know
that the signaling service is not mounting a man-in-the-middle
attack on theor traffic. Even
if they do not use an IdP, as long as
they have minimal trust in the signaling service not to
perform a man-in-the-middle attack, they know that their
communications are secure against the signaling service as
well (i.e., that the signaling service cannot mount a
passive attack on the communications).
</t>
</section>
<section title="Communications and Consent Freshness">
<t>
From a security perspective, everything from here on in is a
little anticlimactic: Alice and Bob exchange data protected by the
keys negotiated by DTLS. Because of the security guarantees discussed
in the previous sections, they know that the communications are
encrypted and authenticated.
</t>
<t>
The one remaining security property we need to establish is
"consent freshness", i.e., allowing Alice to verify that Bob
is still prepared to receive her communications so that Alice
does not continue to send large traffic volumes to entities
which went abruptly offline. ICE
specifies periodic STUN keepalizes but only if media is not flowing.
Because the consent issue is more difficult here, we
require RTCWeb implementations to periodically send keepalives.
As described in Section 5.3, these
keepalives MUST be based on the consent freshness mechanism specified
in <xref target="I-D.muthu-behave-consent-freshness"/>.
If a keepalive fails and no new ICE channels can be established, then
the session is terminated.
</t>
</section>
</section>
<section title="Detailed Technical Description" anchor="sec.proposal.detailed">
<section title="Origin and Web Security Issues" anchor="sec.proposal.origin">
<t>
The basic unit of permissions for RTCWEB is the origin
<xref target="RFC6454"/>. Because the security of the origin
depends on being able to authenticate content from that origin,
the origin can only be securely established if data is transferred
over HTTPS <xref target="RFC2818"/>. Thus, clients MUST treat HTTP and HTTPS origins as
different permissions domains. [Note: this follows directly
from the origin security model and is stated here merely
for clarity.]
</t>
<t>
Many web browsers currently forbid by default any active mixed content
on HTTPS pages. That is, when JavaScript is loaded from an HTTP origin onto
an HTTPS page, an error is displayed and the HTTP content is not
executed unless the user overrides the error. Any browser
which enforces such a policy will also not permit access
to RTCWEB functionality from mixed content pages (because they
never display mixed content). It is
RECOMMENDED that browsers which allow active mixed content
nevertheless disable RTCWEB functionality in mixed content
settings.
[[ OPEN ISSUE: Should this be a 2119 MUST? It's not clear
what set of conditions would make this OK, other than
that browser manufacturers have traditionally been permissive here
here.]]
Note that it is possible for a page which was not mixed content
to become mixed content during the duration of the call.
Implementations MAY choose to terminate the call or display
a warning at that point, but it is also permissible to
ignore this condition. The major risk here is that
the newly arrived insecure JS might redirect media to a location
controlled by the attacker.
This is a deliberate implementation
complexity versus security tradeoff.
[[ OPEN ISSUE:: Should we be more aggressive about this?]]
</t>
</section>
<section title="Device Permissions Model" anchor="sec.proposal.device.permissions">
<t>
Implementations MUST obtain explicit user consent prior to
providing access to the camera and/or microphone. Implementations MUST
at minimum support the following two permissions models for HTTPS
origins.
</t>
<t>
<list style="symbols">
<t>Requests for one-time camera/microphone access.</t>
<t>Requests for permanent access.</t>
</list>
</t>
<t>
Because HTTP origins cannot be securely established against
network attackers, implementations MUST NOT allow the setting
of permanent access permissions for HTTP origins. Implementations
MAY also opt to refuse all permissions grants for HTTP origins,
but it is RECOMMENDED that currently they support one-time
camera/microphone access.
</t>
<t>
In addition, they SHOULD support requests for access that
promise that media from this grant will be sent to a single communicating peer
(obviously there could be other requests for other peers).
E.g., "Call customerservice@ford.com".
The semantics of this request are that the media stream from
the camera and microphone will only be routed through
a connection which has been cryptographically verified
(through the IdP mechanism or an X.509 certificate in the
DTLS-SRTP handshake) as being associated with the stated
identity.
Browsers servicing such requests SHOULD clearly indicate that
identity to the user when asking for permission.
The idea behind this type of permissions is that a user might
have a fairly narrow list of peers he is willing to communicate
with, e.g., "my mother" rather than "anyone on Facebook". Narrow
permissions grants allow the browser to do that enforcement.
</t>
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism for the requesting JS to
indicate which of these forms of permissions it is
requesting. This allows the browser client to know what sort
of user interface experience to provide to the user,
including what permissions to request from the user
and hence what to enforce later.
For instance,
browsers might display a non-invasive door hanger
("some features of this site may not work..." when
asking for long-term permissions) but a more
invasive UI ("here is your own video") for single-call
permissions. The API MAY grant weaker permissions than
the JS asked for if the user chooses to authorize only
those permissions, but if it intends to grant stronger
ones it SHOULD display the appropriate UI for those
permissions and MUST clearly indicate what
permissions are being requested.
</t>
</list></t>
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism for the requesting JS to
relinquish the ability to see or modify the media (e.g., via MediaStream.record()).
Combined with secure authentication of the communicating peer,
this allows a user to be sure that the calling site is not
accessing or modifying their conversion.
</t>
</list></t>
<t><list style="hanging">
<t hangText="UI Requirement:">
The UI MUST clearly indicate when the user's camera
and microphone are in use. This indication MUST NOT be
suppressable by the JS and MUST clearly indicate how to terminate
device access, and provide a UI means to immediately stop camera/microphone
input without the JS being able to prevent it.
</t>
</list></t>
<t><list style="hanging">
<t hangText="UI Requirement:">
If the UI indication of camera/microphone use are displayed
in the browser such that minimizing the browser window would hide the
indication, or the JS creating an overlapping window would hide the
indication, then the browser SHOULD stop camera and microphone input
when the indication is hidden.
[Note: this may not be necessary in systems that are non-windows-based
but that have good notifications support, such as phones.]
</t>
</list>
</t>
<t>
Clients MAY permit the formation of data channels
without any direct user approval. Because sites can always
tunnel data through the server, further restrictions on the
data channel do not provide any additional security.
(though see <xref target="sec.proposal.communications.consent"/>
for a related issue).
</t>
<t>
Implementations which support some form of direct user authentication
SHOULD also provide a policy by which a user can authorize calls
only to specific counterparties. Specifically, the implementation
SHOULD provide the following interfaces/controls:
</t>
<t>
<list style="symbols">
<t>Allow future calls to this verified user.</t>
<t>Allow future calls to any verified user who is in my system address book
(this only works with address book integration, of course).</t>
</list>
</t>
<t>
Implementations SHOULD also provide a different user interface indication
when calls are in progress to users whose identities are directly verifiable.
<xref target="sec.proposal.comsec"/> provides more on this.
</t>
</section>
<section title="Communications Consent" anchor="sec.proposal.communications.consent">
<t>
Browser client implementations of RTCWEB MUST implement ICE.
Server gateway implementations which operate only at public IP
addresses MUST implement either full ICE or ICE-Lite.
</t>
<t>
Browser implementations MUST verify reachability via ICE
prior to sending any non-ICE packets to a given destination.
Implementations MUST NOT provide the ICE transaction ID
to JavaScript during the lifetime of the transaction
(i.e., during the period when the ICE stack would accept
a new response for that transaction). [Note: this document takes no position on
the split between ICE in JS and ICE in the browser. The
above text is written the way it is for editorial convenience and will
be modified appropriately if the WG decides on ICE in the JS.]
The JS MUST NOT be permitted to control the local ufrag and password,
though it of course knows it.
</t>
<t>
While continuing consent is required,
that ICE <xref target="RFC5245"/>; Section 10
keepalives STUN Binding Indications are one-way and
therefore not sufficient.
The current WG consensus is to use ICE Binding Requests
for continuing consent freshness. ICE already requires that
implementations respond to such requests, so this approach is maximally
compatible. A separate document will profile the ICE timers
to be used; see <xref target="I-D.muthu-behave-consent-freshness"/>.
</t>
</section>
<section title="IP Location Privacy" anchor="sec.proposal.ip.location.privacy">
<t>
A side effect of the default
ICE behavior is that the peer learns one's IP address, which leaks
large amounts of location information, especially for mobile
devices. This has
negative privacy consequences in some circumstances. The API requirements
in this section are intended to mitigate this issue. Note that
these requirements are NOT intended to protect the user's IP address
from a malicious site. In general, the site will learn at least a
user's server reflexive address from any HTTP transaction.
Rather, these requirements are intended to allow a site to cooperate
with the user to hide the user's IP address from the other side of the
call. Hiding the user's IP address from the server requires some
sort of explicit privacy preserving mechanism on the client
(e.g., Torbutton [https://www.torproject.org/torbutton/]) and
is out of scope for this specification.
</t>
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism to allow the JS to
suppress ICE negotiation
(though perhaps to allow candidate gathering) until the
user has decided to answer the call [note: determining
when the call has been answered is a question for the JS.]
This enables a user to prevent a peer from learning their
IP address if they elect not to answer a call and also
from learning whether the user is online.
</t>
</list></t>
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism for the calling application JS to
indicate that only TURN candidates are to be used. This
prevents the peer from learning one's IP address at all.
</t>
</list>
</t>
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism for the calling application
to reconfigure an existing call to add non-TURN candidates.
Taken together, this and the previous requirement allow ICE negotiation to start immediately on
incoming call notification, thus reducing post-dial delay, but also to avoid
disclosing the user's IP address until they have
decided to answer. They also allow users to completely hide their IP address
for the duration of the call. Finally, they allow a mechanism for the user to
optimize performance by reconfiguring to allow non-turn candidates during an active call if
the user decides they no longer need to hide their IP address
</t>
</list></t>
</section>
<section title="Communications Security" anchor="sec.proposal.comsec">
<t>
Implementations MUST implement DTLS <xref target="RFC4347"/>
and DTLS-SRTP <xref target="RFC5763"/><xref target="RFC5764"/>. All data
channels MUST be secured via DTLS. DTLS-SRTP MUST be offered
for every media channel and MUST be the default; i.e., if
an implementation receives an offer for DTLS-SRTP and SDES,
DTLS-SRTP MUST be selected. Media traffic MUST NOT be sent
over plain (unencrypted) RTP.
</t>
<t>
[OPEN ISSUE: What should the settings be here? MUST?]
Implementations MAY support SDES for media traffic
for backward compatibility purposes.
</t>
<!-- OPEN ISSUE: DTLS-SRTP key origin scoping? -->
<t><list style="hanging">
<t hangText="API Requirement:">
The API MUST provide a mechanism to indicate that a fresh
DTLS key pair is to be generated for a specific call.
This is intended to allow for unlinkability. Note that
there are also settings where it is attractive to use
the same keying material repeatedly, especially those
with key continuity-based authentication.
</t>
</list></t>
<t><list style="hanging">
<t hangText="API Requirement:">
When DTLS-SRTP is used, the API MUST NOT permit the
JS to obtain the negotiated keying material. This
requirement preserves the end-to-end security of the
media.
</t>
</list></t>
<t><list style="hanging">
<t hangText="UI Requirements: ">
A user-oriented client MUST provide an "inspector" interface which
allows the user to determine the security characteristics of the
media. [largely derived from <xref target="I-D.kaufman-rtcweb-security-ui"/>
</t>
<t>
The following properties SHOULD be displayed "up-front" in the browser
chrome, i.e., without requiring the user to ask for them:
</t>
<t>
<list style="symbols">
<t> A client MUST provide a user interface through which a user may
determine the security characteristics for currently-displayed
audio and video stream(s)</t>
<t> A client MUST provide a user interface through which a user may
determine the security characteristics for transmissions of their
microphone audio and camera video.</t>
<t> The "security characteristics" MUST include an indication as to
whether the cryptographic keys were delivered out-of-band
(from a server) or were generated as a result of a pairwise
negotiation.
</t>
<t>If the far endpoint was directly verified, either via a third-party
verifiable X.509 certificate or via a Web IdP mechanism (see <xref target="sec.generic.idp"/>)
the "security characteristics" MUST include the verified information.
X.509 identities and Web IdP identities
have similar semantics and should be displayed in a similar way.
</t>
</list>
</t>
<t></t>
<t>
The following properties are more likely to require some "drill-down"
from the user:
</t>
<t>
<list style="symbols">
<t> The "security characteristics" MUST indicate the cryptographic algorithms
in use (For example: "AES-CBC" or "Null Cipher".)</t>
<t>The "security characteristics" MUST indicate whether PFS is provided.</t>
<t>The "security characteristics"
MUST include some mechanism to allow an out-of-band verification
of the peer, such as a certificate fingerprint or an SAS.</t>
</list>
</t>
</list></t>
</section>
<section title="Web-Based Peer Authentication" anchor="sec.generic.idp">
<t>
In a number of cases, it is desirable for the endpoint (i.e., the
browser) to be able to directly identity the endpoint on the other
side without trusting only the signaling service to which they
are connected. For instance, users may be making a call via a federated
system where they wish to get direct authentication of the other
side. Alternately, they may be making a call on a site which
they minimally trust (such as a poker site) but to someone who has an identity on
a site they do trust (such as a social network.)
</t>
<t>
Recently, a number of Web-based identity technologies (OAuth, BrowserID, Facebook
Connect), etc. have been developed. While the details vary, what
these technologies share is that they have a Web-based (i.e., HTTP/HTTPS)
identity provider which attests to your identity. For instance,
if I have an account at example.org, I could use the example.org identity
provider to prove to others that I was alice@example.org.
The development of these technologies allows us to separate calling
from identity provision: I could call you on Poker Galaxy but identify
myself as alice@example.org.
</t>
<t>
Whatever the underlying technology, the general principle is that
the party which is being authenticated is NOT the signaling site
but rather the user (and their browser). Similarly, the relying party
is the browser and not the signaling site.
Thus, the browser MUST securely generate the input to the IdP
assertion process and MUST securely display the results of
the verification process to the user in a way which cannot
be imitated by the calling site.
</t>
<t>
The mechanisms defined in this
document do not require the browser to implement any
particular identity protocol or to support any particular
IdP. Instead, this document provides a generic interface
which any IdP can implement. Thus, new IdPs and protocols
can be introduced without change to either the browser or the
calling service. This avoids the need to make a commitment
to any particular identity protocol, although browsers
may opt to directly implement some identity protocols in
order to provide superior performance or UI properties.
</t>
<section title="Trust Relationships: IdPs, APs, and RPs" anchor="sec.trust-relationships">
<t>
Any federated identity protocol has three major participants:
</t>
<t>
<list style="hanging">
<t hangText="Authenticating Party (AP):">The entity which is
trying to establish its identity.</t>
<t></t>
<t hangText="Identity Provider (IdP):">The entity which is
vouching for the AP's identity.</t>
<t></t>
<t hangText="Relying Party (RP):">The entity which is trying
to verify the AP's identity.</t>
</list>
</t>
<t>
The AP and the IdP have an account relationship of some kind: the
AP registers with the IdP and is able to subsequently authenticate
directly to the IdP (e.g., with a password). This means that the
browser must somehow know which IdP(s) the user has
an account relationship with.
This can either be something that the user configures into
the browser or that is configured at the calling site and
then provided to the PeerConnection by the Web application
at the calling site. The use case for having this information
configured into the browser is that the user may "log into"
the browser to bind it to some identity. This is becoming
common in new browsers. However, it should also be possible
for the IdP information to simply be provided by the calling
application.
</t>
<t>
At a high level there are two kinds of IdPs:
</t>
<t>
<list style="hanging">
<t hangText="Authoritative: ">IdPs which have verifiable control
of some section of the identity space. For instance, in the
realm of e-mail, the operator of "example.com" has complete control of the
namespace ending in "@example.com". Thus, "alice@example.com"
is whoever the operator says it is. Examples of systems with
authoritative identity providers include DNSSEC,
RFC 4474, and Facebook Connect (Facebook identities only
make sense within the context of the Facebook system).
</t>
<t></t>
<t hangText="Third-Party: ">IdPs which don't have control of
their section of the identity space but instead verify
user's identities via some unspecified mechanism and then
attest to it. Because the IdP doesn't actually control
the namespace, RPs need to trust that the
IdP is correctly verifying AP identities, and there
can potentially be multiple IdPs attesting to the same
section of the identity space. Probably the best-known example
of a third-party identity provider is SSL certificates,
where there are a large number of CAs all of whom can
attest to any domain name.
</t>
</list>
</t>
<t>
If an AP is authenticating via an authoritative IdP, then
the RP does not need to explicitly configure trust in the IdP at all.
The identity mechanism can directly verify
that the IdP indeed made the relevant identity assertion (a function
provided by the mechanisms in this document), and
any assertion it makes about an identity for which
it is authoritative is directly verifiable. Note that
this does not mean that the IdP might not lie, but that
is a trustworthiness judgement that the user can make
at the time he looks at the identity.
</t>
<t>
By contrast, if an AP is authenticating via a third-party
IdP, the RP needs to explicitly trust that IdP
(hence the need for an explicit trust anchor list
in PKI-based SSL/TLS clients). The list of trustable
IdPs needs to be configured directly into the
browser, either by the user or potentially by the
browser manufacturer. This is a significant advantage
of authoritative IdPs and implies that if third-party
IdPs are to be supported, the potential number needs
to be fairly small.
</t>
</section>
<section title="Overview of Operation" anchor="sec.overview">
<t>
In order to provide security without trusting the calling
site, the PeerConnection component of the browser must
interact directly with the IdP. The details of the
mechanism are described in the W3C API specification,
but the general idea is that the PeerConnection component
downloads JS from a specific location on the
IdP dictated by the IdP domain name. That JS (the "IdP proxy")
runs in
an isolated security context within the browser
and the PeerConnection
talks to it via a secure message passing channel.
</t>
<t>
Note that there are two logically separate functions here:
</t>
<t>
<list style="symbols">
<t>Identity assertion generation.</t>
<t>Identity assertion verification.</t>
</list>
</t>
<t>
The same IdP JS "endpoint" is used for both functions but
of course a given IdP might behave differently and load
new JS to perform one function or the other.
</t>
<figure>
<artwork><![CDATA[
+------------------------------------+
| https://calling-site.example.com |
| |
| |
| |
| Calling JS Code |
| ^ |
| | API Calls |
| v |
| PeerConnection |
| ^ |
| | postMessage() |
| v |
| +-------------------------+ | +---------------+
| | https://idp.example.org | | | |
| | |<--------->| Identity |
| | IdP JS | | | Provider |
| | | | | |
| +-------------------------+ | +---------------+
| |
+------------------------------------+
]]></artwork>
</figure>
<t>
When the PeerConnection object wants to interact with the
IdP, the sequence of events is as follows:
</t>
<t>
<list style="numbers">
<t>The browser (the PeerConnection component)
instantiates an IdP proxy with its source at the IdP. This allows the
IdP to load whatever JS is necessary into the
proxy, which runs in the IdP's security context.</t>
<t>If the user is not already logged in, the
IdP does whatever is required to log them in,
such as soliciting a username and password.</t>
<t>Once the user is logged in, the IdP proxy
notifies the browser that
it is ready.</t>
<t>The browser and the IdP proxy communicate
via a standardized series of messages
delivered via postMessage. For instance,
the browser might request the IdP proxy to
sign or verify a given identity assertion.</t>
</list>
</t>
<t>
This approach allows us to decouple the browser from
any particular identity provider; the browser need
only know how to load the IdP's JavaScript--which
is deterministic from the IdP's identity--and
the generic protocol for requesting and verifying
assertions. The IdP provides whatever logic
is necessary to bridge the generic protocol to
the IdP's specific requirements. Thus, a single
browser can support any number of identity protocols,
including being forward compatible with IdPs which
did not exist at the time the browser was written.
</t>
</section>
<section title="Items for Standardization" anchor="sec.standardized">
<t>
In order to make this work, we must standardize the following
items:
</t>
<t>
<list style="symbols">
<t>
The precise information from the signaling message that
must be cryptographically bound to the user's identity
and a mechanism for carrying assertions in JSEP
messages. <xref target="sec.jsep-binding"/>
</t>
<t>
The interface to the IdP. <xref target="sec.protocol-details"/>
specifies a specific protocol mechanism which allows the
use of any identity protocol without requiring specific
further protocol support in the browser
</t>
<t>
The JavaScript interfaces which the calling application can
use to specify the IdP to use to generate assertions and
to discover what assertions were received.
</t>
</list>
</t>
<t>
The first two items are defined in this document. The final one
is defined in the companion W3C WebRTC API specification [TODO:REF]
</t>
</section>
<section title="Binding Identity Assertions to JSEP Offer/Answer Transactions" anchor="sec.jsep-binding">
<section title="Input to Assertion Generation Process">
<t>
As discussed above, an identity assertion binds the user's
identity (as asserted by the IdP) to the JSEP offer/exchange
transaction and specifically to the media. In order to
achieve this, the PeerConnection must provide the DTLS-SRTP
fingerprint to be bound to the identity. This is provided
in a JSON structure for extensibility, as shown below:
</t>
<figure>
<artwork><![CDATA[
{
"fingerprint" :
{
"algorithm":"SHA-1",
"digest":"4A:AD:B9:B1:3F:...:E5:7C:AB"
}
}
]]>
</artwork>
</figure>
<t>
The "algorithm" and digest values correspond directly to the
algorithm and digest in the a=fingerprint line of the SDP.
</t>
<t>
Note: this structure does not need to be interpreted by the
IdP or the IdP proxy. It is consumed solely by the RP's browser.
The IdP merely treats it as an opaque value to be
attested to.
Thus, new parameters can be added to the assertion without
modifying the IdP.
</t>
</section>
<section title="Carrying Identity Assertions">
<t>
Once an IdP has generated an assertion, it is attached to
the JSEP message. This is done by adding a new a-line to the
SDP, of the form a=identity. The sole contents of this value
are a base-64-encoded version of the identity assertion.
For example:
</t>
<figure>
<artwork><![CDATA[
v=0
o=- 1181923068 1181923196 IN IP4 ua1.example.com
s=example1
c=IN IP4 ua1.example.com
a=setup:actpass
a=fingerprint: SHA-1 \
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
a=identity: \
ImlkcCI6eyJkb21haW4iOiAiZXhhbXBsZS5vcmciLCAicHJvdG9jb2wiOiAiYm9n \
dXMifSwiYXNzZXJ0aW9uIjpcIntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5v \
cmdcIixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIs \
XCJzaWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9Cg==
t=0 0
m=audio 6056 RTP/AVP 0
a=sendrecv
a=tcap:1 UDP/TLS/RTP/SAVP RTP/AVP
a=pcfg:1 t=1
]]>
</artwork>
</figure>
<t>
Each identity attribute should be paired (and attests to) with an a=fingerprint
attribute and therefore can exist either at the session or media level.
Multiple identity attributes may appear at either level, though it is
RECOMMENDED that implementations
not do this, because it becomes very unclear what security claim that
they are making and the UI guidelines above become unclear. Browsers
MAY choose refuse to display any identity indicators in the face
of multiple identity attributes with different identities
but SHOULD process multiple attributes with the same identity
as described above.
</t>
</section>
</section>
<section title="IdP Interaction Details" anchor="sec.protocol-details">
<section title="General Message Structure">
<t>
Messages between the PeerConnection object and the
IdP proxy are formatted using JSON <xref target="RFC4627"/>.
For instance, the PeerConnection would request a
signature with the following "SIGN" message:
</t>
<figure>
<artwork><![CDATA[
{
"type":"SIGN",
"id": "1",
"origin":"https://calling-site.example.com",
"message":"012345678abcdefghijkl"
}
]]></artwork>
</figure>
<t>
All messages MUST contain a "type" field which indicates
the general meaning of the message.
</t>
<t>
All requests from the PeerConnection object MUST contain
an "id" field which MUST be unique for that PeerConnection
object. Any responses from the IdP proxy MUST contain
the same id in response, which allows the PeerConnection
to correlate requests and responses.
</t>
<t>
All requests from the PeerConnection object MUST contain
an "origin" field containing the origin of the JS which
initiated the PC (i.e., the URL of the calling site).
This origin value can be used by the IdP to make access
control decisions. For instance, an IdP might
only issue identity assertions for certain calling
services in the same way that some IdPs require that
relying Web sites have an API key before learning user
identity.
</t>
<t>
Any message-specific data is carried in a "message"
field. Depending on the message type, this may
either be a string or a richer JSON object.
</t>
<section title="Errors">
<t>
If an error occurs, the IdP sends a message of
type "ERROR". The message MAY have an
"error" field containing freeform
text data which containing additional
information about what happened. For instance:
</t>
<figure title="Example error" anchor="fig.example-error">
<artwork><![CDATA[
{
"type":"ERROR",
"error":"Signature verification failed"
}
]]></artwork>
</figure>
</section>
</section>
<section title="IdP Proxy Setup" anchor="sec.iframe-setup">
<t>
In order to perform an identity transaction, the PeerConnection
must first create an IdP proxy. While the details
of this are specified in the W3C API document, from the
perspective of this specification, however, the relevant
facts are:
</t>
<t>
<list style="symbols">
<t>The JS runs in the IdP's security context with the
base page retrieved from the URL specified in
<xref target="sec.idp-uri"/></t>
<t>The usual browser sandbox isolation mechanisms MUST
be enforced with respect to the IdP proxy.</t>
<t>JS running in the IdP proxy MUST be able to
send and receive messages to the PeerConnection
and the PC and IdP proxy
are able to verify the source and destination of
these messages.</t>
</list>
</t>
<t>
Initially the IdP proxy is in an unready state; the
IdP JS must be loaded and there may be several round
trips to the IdP server, for instance to log the user
in. When the IdP proxy is ready to receive commands,
it delivers a "ready" message. As this
message is unsolicited, it simply contains:
</t>
<figure>
<artwork><![CDATA[
{ "type":"READY" }
]]></artwork>
</figure>
<t>
[[ OPEN ISSUE: if the W3C half of this converges on WebIntents, then
the READY message will not be necessary.]]</t>
<t>
Once the PeerConnection object receives the ready message,
it can send commands to the IdP proxy.
</t>
<section title="Determining the IdP URI" anchor="sec.idp-uri">
<t>
In order to ensure that the IdP is under control of the
domain owner rather than someone who merely has an account
on the domain owner's server (e.g., in shared hosting
scenarios), the IdP JavaScript is hosted at a deterministic
location based on the IdP's domain name.
Each IdP proxy instance is associated with two values:
</t>
<t>
<list style="hanging">
<t hangText="domain name:">The IdP's domain name</t>
<t hangText="protocol:">The specific IdP protocol
which the IdP is using. This is a completely IdP-specific
string, but allows an IdP to implement two protocols
in parallel. This value may be the empty string.</t>
</list>
</t>
<t>
Each IdP MUST serve its initial entry page
(i.e., the one loaded by the IdP proxy) from
the well-known URI specified in
"/.well-known/idp-proxy/<protocol>" on
the IdP's web site. This URI MUST be loaded
via HTTPS <xref target="RFC2818"/>.
For example, for the IdP "identity.example.com"
and the protocol "example", the URL
would be:
</t>
<figure>
<artwork><![CDATA[
https://example.com/.well-known/idp-proxy/example
]]></artwork>
</figure>
<section title="Authenticating Party">
<t>
How an AP determines the appropriate IdP domain is
out of scope of this specification. In general,
however, the AP has some actual account relationship
with the IdP, as this identity is what the IdP is
attesting to. Thus, the AP somehow supplies the
IdP information to the browser. Some potential
mechanisms include:
</t>
<t>
<list style="symbols">
<t>Provided by the user directly.</t>
<t>Selected from some set of IdPs known to the calling site.
E.g., a button that shows "Authenticate via Facebook Connect"</t>
</list>
</t>
</section>
<section title="Relying Party">
<t>
Unlike the AP, the RP need not have any particular
relationship with the IdP. Rather, it needs to be able
to process whatever assertion is provided by the AP.
As the assertion contains the IdP's identity,
the URI can be constructed directly from the
assertion, and thus the RP can directly verify
the technical validity of the assertion with no user
interaction. Authoritative assertions need only
be verifiable. Third-party assertions also MUST be
verified against local policy, as described in
<xref target="sec.id-format"/>.
</t>
</section>
</section>
<section title="Requesting Assertions" anchor="sec.request-assert">
<t>
In order to request an assertion, the PeerConnection
sends a "SIGN" message. Aside from the mandatory fields,
this message has a "message" field containing a string.
The contents of this string are defined above, but are
opaque from the perspective of the IdP.
</t>
<t>
A successful response to a "SIGN" message contains
a message field which is a JS dictionary
dictionary consisting of two fields:
</t>
<t>
<list style="hanging">
<t hangText="idp:">A dictionary containing the domain name of the provider
and the protocol string</t>
<t hangText="assertion:">An opaque field containing the
assertion itself. This is only interpretable by the
idp or its proxy.</t>
</list>
</t>
<t>
<xref target="fig.assert-request"/> shows an
example transaction, with the message
"abcde..." (remember, the messages are opaque at this layer) being signed and bound to identity
"ekr@example.org". In this case, the message has presumably
been
digitally signed/MACed in some way that the IdP can
later verify it, but this is an implementation detail
and out of scope of this document. Line breaks are inserted
solely for readability.</t>
<figure title="Example assertion request" anchor="fig.assert-request">
<artwork><![CDATA[
PeerConnection -> IdP proxy:
{
"type":"SIGN",
"id":1,
"origin":"https://calling-service.example.com/",
"message":"abcdefghijklmnopqrstuvwyz"
}
IdPProxy -> PeerConnection:
{
"type":"SUCCESS",
"id":1,
"message": {
"idp":{
"domain": "example.org"
"protocol": "bogus"
},
"assertion":\"{\"identity\":\"bob@example.org\",
\"contents\":\"abcdefghijklmnopqrstuvwyz\",
\"request_origin\":\"rtcweb://peerconnection\",
\"signature\":\"010203040506\"}"
}
}
]]>
</artwork>
</figure>
</section>
<section title="Verifying Assertions" anchor="sec.verify-assert">
<t>
In order to verify an assertion, an RP sends a "VERIFY"
message to the IdP proxy containing the assertion
supplied by the AP in the "message" field.
</t>
<t>
The IdP proxy verifies the assertion. Depending on the
identity protocol, this may require one or more round
trips to the IdP. For instance, an OAuth-based protocol
will likely require using the IdP as an oracle,
whereas with BrowserID the IdP proxy can likely
verify the signature on the assertion without
contacting the IdP, provided that it has cached
the IdP's public key.
</t>
<t>
Regardless of the mechanism,
if verification succeeds, a successful response from
the IdP proxy MUST contain a message field consisting
of a dictionary/hash with the following fields:
</t>
<t>
<list style="hanging">
<t hangText="identity">The identity of the AP from the
IdP's perspective. Details of this are provided in
<xref target="sec.id-format"/></t>
<t hangText="contents">The original unmodified string
provided by the AP in the original SIGN request.</t>
<t hangText="request_origin">The original origin of
the SIGN request on the AP side as determined by
the origin of the PostMessage call. The IdP MUST somehow
arrange to propagate this information as part of the
assertion. The receiving PeerConnection MUST verify that
this value is "rtcweb://peerconnection" (which implies
that PeerConnection must arrange that its messages to the
IdP proxy are from this origin.)
[[ OPEN ISSUE: Can a URI person help make a better URI.]]
</t>
</list>
</t>
<t>
<xref target="fig.verify-request"/> shows
an example transaction. Line breaks are inserted
solely for readability.
</t>
<figure title="Example verification request" anchor="fig.verify-request">
<artwork>
<![CDATA[
PeerConnection -> IdP Proxy:
{
"type":"VERIFY",
"id":2,
"origin":"https://calling-service.example.com/",
"message":\"{\"identity\":\"bob@example.org\",
\"contents\":\"abcdefghijklmnopqrstuvwyz\",
\"request_origin\":\"rtcweb://peerconnection\",
\"signature\":\"010203040506\"}"
}
IdP Proxy -> PeerConnection:
{
"type":"SUCCESS",
"id":2,
"message": {
"identity" : {
"name" : "bob@example.org",
"displayname" : "Bob"
},
"request_origin":"rtcweb://peerconnection",
"contents":"abcdefghijklmnopqrstuvwyz"
}
}
]]>
</artwork>
</figure>
<section title="Identity Formats" anchor="sec.id-format">
<t>
Identities passed from the IdP proxy to the PeerConnection
are structured as JSON dictionaries with one mandatory
field: "name". This field MUST consist of an
RFC822-formatted string representing the user's identity.
[[ OPEN ISSUE: Would it be better to have a typed field? ]]
The PeerConnection API MUST check this string as
follows:
</t>
<t>
<list style="numbers">
<t>If the RHS of the string is equal to the domain name
of the IdP proxy, then the assertion is valid,
as the IdP is authoritative for this domain.</t>
<t>If the RHS of the string is not equal to the domain
name of the IdP proxy, then the PeerConnection object
MUST reject the assertion unless (a) the
IdP domain is listed as an acceptable third-party
IdP and (b) local policy is configured to trust
this IdP domain for the RHS of the identity string.
</t>
</list>
</t>
<t>
Sites which have identities that do not fit
into the RFC822 style (for instance, Facebook
ids are simple numeric values) SHOULD
convert them to this form by appending their
IdP domain (e.g., 12345@identity.facebook.com),
thus ensuring that they are authoritative for
the identity.
</t>
<t>
The IdP proxy MAY also include a "displayname" field
which contains a more user-friendly identity assertion.
Browsers SHOULD take care in the UI to distinguish the
"name" assertion which is verifiable directly from the
"displayname" which cannot be verified and thus
relies on trust in the IdP. In future,
we may define other fields to allow the IdP to
provide more information to the browser.
[[OPEN ISSUE: Should this field exist? Is it
confusing? ]]
</t>
</section>
</section>
</section>
</section>
</section>
<section title="Security Considerations" anchor="sec.sec-cons">
<t>
Much of the security analysis of this problem is contained in
<xref target="I-D.ietf-rtcweb-security"/> or in the discussion
of the particular issues above. In order to avoid
repetition, this section focuses on (a) residual threats
that are not addressed by this document and (b) threats
produced by failure/misbehavior of one of the components
in the system.
</t>
<section title="Communications Security">
<t>
While this document favors DTLS-SRTP, it permits a variety
of communications security mechanisms and thus the level
of communications security actually provided varies
considerably. Any pair of implementations which have
multiple security mechanisms in common are subject to
being downgraded to the weakest of those common
mechanisms by any attacker who can modify the signaling
traffic. If communications are over HTTP, this means
any on-path attacker. If communications are over HTTPS,
this means the signaling server. Implementations which
wish to avoid downgrade attack should only offer
the strongest available mechanism, which is DTLS/DTLS-SRTP.
Note that the implication of this choice will be that
interop to non-DTLS-SRTP devices will need to happen through
gateways.
</t>
<t>
Even if only DTLS/DTLS-SRTP are used, the signaling server
can potentially mount a man-in-the-middle attack unless
implementations have some mechanism for independently
verifying keys. The UI requirements in <xref target="sec.proposal.comsec"/>
are designed to provide such a mechanism for motivated/security
conscious users, but are not suitable for general use.
The identity service mechanisms in <xref target="sec.generic.idp"/>
are more suitable for general use. Note, however, that
a malicious signaling service can strip off any such
identity assertions, though it cannot forge new ones.
Note that all of the third-party security mechanisms available
(whether X.509 certificates or a third-party IdP) rely on
the security of the third party--this is of course also true
of your connection to the Web site itself. Users who wish to
assure themselves of security against a malicious identity
provider can only do so by verifying peer credentials directly, e.g., by
checking the peer's fingerprint against a value delivered out
of band.
</t>
<t>
In order to protect against malicious content JavaScript, that
JavaScript MUST NOT be allowed to have direct access to---or perform
computations with---DTLS keys. For instance, if content JS were
able to compute digital signatures, then it would be possible
for content JS to get an identity assertion for a browser's
generated key and then use that assertion plus a signature by
the key to authenticate a call protected under an ephemeral
DH key controlled by the content JS, thus violating the security
guarantees otherwise provided by the IdP mechanism. Note that
it is not sufficient merely to deny the content JS direct access to
the keys, as some have suggested doing with the WebCrypto API.
The JS must also not be allowed to perform operations that would
be valid for a DTLS endpoint. By far the safest approach is simply
to deny the ability to perform any operations that depend on
secret information associated with the key. Operations that depend
on public information, such as exporting the public key are of course
safe.
</t>
</section>
<section title="Privacy">
<t>
The requirements in this document are intended to allow:
</t>
<t>
<list style="symbols">
<t>Users to participate in calls without revealing their location.</t>
<t>Potential callees to avoid revealing their location and even
presence status prior to agreeing to answer a call.</t>
</list>
</t>
<t>
However, these privacy protections come at a performance cost
in terms of using TURN relays and, in the latter case, delaying
ICE. Sites SHOULD make users aware of these tradeoffs.
</t>
<t>
Note that the protections provided here assume a non-malicious
calling service. As the calling service always knows the users
status and (absent the use of a technology like Tor) their
IP address, they can violate the users privacy at will.
Users who wish privacy against the calling sites they
are using must use separate privacy enhancing technologies
such as Tor. Combined RTCWEB/Tor implementations SHOULD
arrange to route the media as well as the signaling through
Tor. [Currently this will produce very suboptimal performance.]
</t>
</section>
<section title="Denial of Service">
<t>
The consent mechanisms described in this document are intended to
mitigate denial of service attacks in which an attacker uses
clients to send large amounts of traffic to a victim without
the consent of the victim. While these mechanisms are sufficient
to protect victims who have not implemented RTCWEB at all,
RTCWEB implementations need to
be more careful.
</t>
<t>
Consider the case of a call center which accepts calls via
RTCWeb. An attacker proxies the call center's front-end
and arranges for multiple clients to initiate calls to
the call center. Note that this requires user consent
in many cases but because the data channel does not need
consent, he can use that directly. Since ICE will
complete, browsers can then be induced to send large
amounts of data to the victim call center if it supports
the data channel at all. Preventing
this attack requires that automated RTCWEB implemementations
implement sensible flow control and have the ability to
triage out (i.e., stop responding to ICE probes on)
calls which are behaving badly, and especially to
be prepared to remotely throttle the data channel in
the absence of plausible audio and video (which
the attacker cannot control).
</t>
<t>
Another related attack is for the signaling service to
swap the ICE candidates for the audio and video streams,
thus forcing a browser to send video to the sink that
the other victim expects will contain audio
(perhaps it is only expecting audio!)
potentially causing overload.
Muxing multiple media flows over a single transport makes
it harder to individually suppress a single flow by denying
ICE keepalives. Either media-level (RTCP) mechanisms must be
used or the implementation must deny responses entirely,
thus termnating the call.
</t>
<t>
Yet another attack, suggested by Magnus Westerlund,
is for the attacker to cross-connect offers and answers
as follows. It induces the victim to make a call
and then uses its control of other users browsers
to get them to attempt a call to someone. It then
translates their offers into apparent answers to
the victim, which looks like large-scale parallel forking.
The victim still responds to ICE responses and
now the browsers all try to send media to the victim.
Implementations can defend themselves from this attack
by only responding to ICE Binding Requests for a limited
number of remote ufrags (this is the reason for
the requirement that the JS not be able to control
the ufrag and password).
</t>
<t>
Note that attacks based on confusing one end or the other about
consent are possible even in the face of the
third-party identity mechanism as long as major parts of
the signaling messages are not signed. On the other hand,
signing the entire message severely restricts the capabilities
of the calling application, so there are difficult tradeoffs here.
</t>
</section>
<section title="IdP Authentication Mechanism">
<t>
This mechanism relies for its security on the IdP and on
the PeerConnection correctly enforcing the security
invariants described above. At a high level, the IdP
is attesting that the user identified in the assertion
wishes to be associated with the assertion. Thus,
it must not be possible for arbitrary third parties to
get assertions tied to a user or to produce assertions
that RPs will accept.
</t>
<section title="PeerConnection Origin Check" anchor="sec.pc-origin">
<t>
Fundamentally, the IdP proxy is just a piece of HTML and JS
loaded by the browser, so nothing stops a Web attacker o
from creating their own IFRAME, loading the IdP proxy HTML/JS,
and requesting a signature. In order to prevent this attack,
we require that all signatures be tied to a specific
origin ("rtcweb://...") which cannot be produced by
content JavaScript. Thus, while an attacker
can instantiate the IdP proxy, they cannot send messages
from an appropriate origin and so cannot create acceptable
assertions. I.e., the assertion request must have come
from the browser. This origin check is enforced on the relying
party side, not on the authenticating party side. The reason
for this is to take the burden of knowing which origins
are valid off of the IdP, thus making this mechanism
extensible to other applications besides RTCWEB. The
IdP simply needs to gather the origin information
(from the posted message) and attach it to the assertion.
</t>
<t>
Note that although this origin check is enforced on the RP
side and not at the IdP, it is absolutely imperative that
it be done. The mechanisms in this document rely on the
browser enforcing access restrictions on the DTLS keys
and assertion requests which do not come with the right
origin may be from content JS rather than from browsers,
and therefore those access restrcitions cannot be assumed.
</t>
<t>
Note that this check only asserts that the browser (or some
other entity with access to the user's authentication data)
attests to the request and hence to the fingerprint.
It does not demonstrate that the browser has access to the
associated private key. However, attaching one's
identity to a key that the user does not control does
not appear to provide substantial leverage to an attacker,
so
a proof of possession is omitted for simplicity.
</t>
</section>
<section title="IdP Well-known URI" anchor="sec.sec-idp-uri">
<t>
As described in <xref target="sec.idp-uri"/> the IdP proxy
HTML/JS landing page is located at a well-known URI based on
the IdP's domain name. This requirement prevents an attacker
who can write some resources at the IdP (e.g., on one's
Facebook wall) from being able to impersonate the IdP.
</t>
</section>
<section title="Privacy of IdP-generated identities and the hosting site">
<t>
Depending on the structure of the IdP's assertions, the calling
site may learn the user's identity from the perspective of the IdP.
In many cases this is not an issue because the user is authenticating
to the site via the IdP in any case, for instance when the user
has logged in with Facebook Connect and is then authenticating their
call with a Facebook identity.
However, in other case, the user
may not have already revealed their identity to the site.
In general, IdPs SHOULD either verify that the user is willing
to have their identity revealed to the site (e.g., through
the usual IdP permissions dialog) or arrange that the identity
information is only available to known RPs (e.g., social graph
adjacencies) but not to the calling site. The "origin" field
of the signature request can be used to check that the
user has agreed to disclose their identity to the calling
site; because it is supplied by the PeerConnection it can
be trusted to be correct.
</t>
</section>
<section title="Security of Third-Party IdPs" anchor="sec.sec-third-party">
<t>
As discussed above, each third-party IdP represents a new universal trust
point and therefore the number of these IdPs needs to be
quite limited. Most IdPs, even those which issue unqualified
identities such as Facebook, can be recast as authoritative
IdPs (e.g., 123456@facebook.com). However, in such cases,
the user interface implications are not entirely desirable.
One intermediate approach is to have special (potentially user
configurable) UI for large authoritative IdPs, thus allowing
the user to instantly grasp that the call is being authenticated
by Facebook, Google, etc.
</t>
</section>
<section title="Web Security Feature Interactions">
<t>
A number of optional Web security features have the potential
to cause issues for this mechanism, as discussed below.
</t>
<section title="Popup Blocking" anchor="sec.popup-blocking">
<t>
If the user is not already logged into the IdP, the
IdP proxy may need to pop up a top level window in order
to prompt the user for their authentication information
(it is bad practice to do this in an IFRAME inside the
window because then users have no way to determine the
destination for their password). If the user's browser
is configured to prevent popups, this may fail
(depending on the exact algorithm that the popup blocker
uses to suppress popups). It may be necessary to provide
a standardized mechanism to allow the IdP proxy to
request popping of a login window. Note that
care must be taken here to avoid PeerConnection becoming
a general escape hatch from popup blocking. One possibility
would be to only allow popups when the user has explicitly
registered a given IdP as one of theirs (this is only relevant
at the AP side in any case). This is what WebIntents does, and
the problem would go away if WebIntents is used.
</t>
</section>
<section title="Third Party Cookies" anchor="sec.3rd-party-cookies">
<t>
Some browsers allow users to block third party cookies (cookies associated
with origins other than the top level page) for privacy reasons.
Any IdP which uses cookies to persist logins will be broken
by third-party cookie blocking. One option is to accept this
as a limitation; another is to have the PeerConnection object
disable third-party cookie blocking for the IdP proxy.
</t>
</section>
</section>
</section>
</section>
</section>
<section title="Acknowledgements">
<t>
Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta,
Cullen Jennings, Hadriel Kaplan, Matthew Kaufman,
Jim McEachern, Martin Thomson, Magnus Westerland.
</t>
</section>
<section title="Changes">
<section title="Changes since -05">
<t>
The following changes have been made since the -05 draft.
</t>
<t>
<list style="symbols">
<t>Response to comments from Richard Barnes</t>
<t>More explanation of the IdP security properties and the
federation use case.</t>
<t>Editorial cleanup.</t>
</list>
</t>
</section>
<section title="Changes since -03">
<t>Version -04 was a version control mistake. Please ignore.</t>
<t>
The following changes have been made since the -04 draft.
</t>
<t>
<list style="symbols">
<t>Move origin check from IdP to RP per discussion in YVR.</t>
<t>Clarified treatment of X.509-level identities.</t>
<t>Editorial cleanup.</t>
</list>
</t>
</section>
<section title="Changes since -03">
</section>
<section title="Changes since -02">
<t>
The following changes have been made since the -02 draft.
</t>
<t>
<list style="symbols">
<t>Forbid persistent HTTP permissions.</t>
<t>Clarified the text in S 5.4 to clearly refer to requirements on the API to provide functionality to the site.</t>
<t>Fold in the IETF portion of draft-rescorla-rtcweb-generic-idp</t>
<t>Retarget the continuing consent section to assume Binding Requests</t>
<t>Editorial improvements</t>
</list>
</t>
</section>
</section>
</middle>
<back>
<references title="Normative References">
&RFC2119;
&RFC2818;
&RFC6454;
&RFC5245;
&RFC4347;
&RFC5763;
&RFC5764;
&RFC4627;
&I-D.ietf-rtcweb-security;
&I-D.muthu-behave-consent-freshness;
</references>
<references title="Informative References">
&RFC3261;
&RFC5705;
&RFC6455;
&I-D.kaufman-rtcweb-security-ui;
&I-D.jennings-rtcweb-signaling;
&I-D.ietf-rtcweb-jsep;
<reference anchor="XmlHttpRequest">
<front>
<title>XMLHttpRequest Level 2</title>
<author initials="A." surname="van Kesteren">
<organization></organization>
</author>
</front>
<format target="http://www.w3.org/TR/XMLHttpRequest/" type="TXT"/>
</reference>
</references>
<section title="Example IdP Bindings to Specific Protocols">
<t>
This section provides some examples of how the
mechanisms described in this document could be used
with existing authentication protocols such as
BrowserID or OAuth. Note that this does not
require browser-level support for either protocol.
Rather, the protocols can be fit into the generic
framework. (Though BrowserID in particular works
better with some client side support).
</t>
<section title="BrowserID">
<t>
BrowserID [https://browserid.org/] is a technology which
allows a user with a verified email address to generate
an assertion (authenticated by their identity provider)
attesting to their identity (phrased as an email address).
The way that this is used in practice is that the relying
party embeds JS in their site which talks to the BrowserID
code (either hosted on a trusted intermediary or embedded
in the browser). That code generates the assertion which is
passed back to the relying party for verification.
The assertion can be verified directly or with a Web
service provided by the identity provider.
It's relatively easy to extend this functionality to
authenticate RTCWEB calls, as shown below.
</t>
<figure>
<artwork><![CDATA[
+----------------------+ +----------------------+
| | | |
| Alice's Browser | | Bob's Browser |
| | OFFER ------------> | |
| Calling JS Code | | Calling JS Code |
| ^ | | ^ |
| | | | | |
| v | | v |
| PeerConnection | | PeerConnection |
| | ^ | | | ^ |
| Finger| |Signed | |Signed | | |
| print | |Finger | |Finger | |"Alice"|
| | |print | |print | | |
| v | | | v | |
| +--------------+ | | +---------------+ |
| | IdP Proxy | | | | IdP Proxy | |
| | to | | | | to | |
| | BrowserID | | | | BrowserID | |
| | Signer | | | | Verifier | |
| +--------------+ | | +---------------+ |
| ^ | | ^ |
+-----------|----------+ +----------|-----------+
| |
| Get certificate |
v | Check
+----------------------+ | certificate
| | |
| Identity |/-------------------------------+
| Provider |
| |
+----------------------+
]]></artwork>
</figure>
<t>
The way this mechanism works is as follows. On Alice's side, Alice
goes to initiate a call.
</t>
<t><list style="numbers">
<t>The calling JS instantiates a PeerConnection
and tells it that it is interested in having it authenticated
via BrowserID (i.e., it provides "browserid.org" as the
IdP name.)</t>
<t>The PeerConnection instantiates the BrowserID signer in
the IdP proxy
</t>
<t>The BrowserID signer contacts Alice's identity provider,
authenticating as Alice (likely via a cookie).</t>
<t>The identity provider returns a short-term certificate
attesting to Alice's identity and her short-term public key.</t>
<t>The Browser-ID code signs the fingerprint and returns the
signed assertion + certificate to the PeerConnection. </t>
<t>The PeerConnection returns the signed information to the
calling JS code.</t>
<t>The signed assertion gets sent over the wire to Bob's
browser (via the signaling service) as part of the call setup.</t>
</list>
</t>
<t>
Obviously, the format of the signed assertion varies depending
on what signaling style the WG ultimately adopts. However, for
concreteness, if something like ROAP were adopted, then the
entire message might look like:
</t>
<figure>
<artwork><![CDATA[
{
"messageType":"OFFER",
"callerSessionId":"13456789ABCDEF",
"seq": 1
"sdp":"
v=0\n
o=- 2890844526 2890842807 IN IP4 192.0.2.1\n
s= \n
c=IN IP4 192.0.2.1\n
t=2873397496 2873404696\n
m=audio 49170 RTP/AVP 0\n
a=fingerprint: SHA-1 \
4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\n",
"identity":{
"idp":{ // Standardized
"domain":"browserid.org",
"method":"default"
},
"assertion": // Contents are browserid-specific
"\"assertion\": {
\"digest\":\"<hash of the contents from the browser>\",
\"audience\": \"[TBD]\"
\"valid-until\": 1308859352261,
},
\"certificate\": {
\"email\": \"rescorla@example.org\",
\"public-key\": \"<ekrs-public-key>\",
\"valid-until\": 1308860561861,
}" // certificate is signed by example.org
}
}
]]></artwork>
</figure>
<t>
<!-- [TODO: Need to talk about Audience a bit.] -->
Note that while the IdP here is specified as "browserid.org",
the actual certificate is signed by example.org. This is because
BrowserID is a combined authoritative/third-party system in
which browserid.org delegates the right to be authoritative
(what BrowserID calls primary) to individual domains.
</t>
<t>
On Bob's side, he receives the signed assertion as part of the call
setup message and a similar procedure happens to verify it.
</t>
<t><list style="numbers">
<t>The calling JS instantiates a PeerConnection
and provides it the relevant signaling information, including the
signed assertion.</t>
<t>The PeerConnection instantiates the IdP proxy which examines
the IdP name and brings up the BrowserID verification code.</t>
<t>The BrowserID verifier contacts the identity provider to
verify the certificate and then uses the key to verify the
signed fingerprint.</t>
<t>Alice's verified identity is returned to the PeerConnection
(it already has the fingerprint).</t>
<t>At this point, Bob's browser can display a trusted UI indication
that Alice is on the other end of the call.</t>
</list>
</t>
<t>
When Bob returns his answer, he follows the converse procedure, which
provides Alice with a signed assertion of Bob's identity and keying
material.
</t>
</section>
<section title="OAuth">
<t>
While OAuth is not directly designed for user-to-user authentication,
with a little lateral thinking it can be made to serve. We use the
following mapping of OAuth concepts to RTCWEB concepts:
</t>
<texttable anchor="oauth-rtcweb">
<ttcol align="left">OAuth</ttcol>
<ttcol align="left">RTCWEB</ttcol>
<c>Client</c><c>Relying party</c>
<c>Resource owner</c><c>Authenticating party</c>
<c>Authorization server</c><c>Identity service</c>
<c>Resource server</c><c>Identity service</c>
</texttable>
<t>
The idea here is that when Alice wants to authenticate to Bob (i.e., for
Bob to be aware that she is calling). In order to do this, she allows
Bob to see a resource on the identity provider that is bound to the
call, her identity, and her public key. Then Bob retrieves the resource
from the identity provider, thus verifying the binding between Alice
and the call.
</t>
<figure>
<artwork><![CDATA[
Alice IdP Bob
---------------------------------------------------------
Call-Id, Fingerprint ------->
<------------------- Auth Code
Auth Code ---------------------------------------------->
<----- Get Token + Auth Code
Token --------------------->
<------------- Get call-info
Call-Id, Fingerprint ------>
]]></artwork>
</figure>
<t>
This is a modified version of a common OAuth flow, but
omits the redirects required to have the client point the
resource owner to the IdP, which is acting as both
the resource server and the authorization server, since
Alice already has a handle to the IdP.
</t>
<t>
Above, we have referred to "Alice", but really what we mean
is the PeerConnection. Specifically, the PeerConnection will
instantiate an IFRAME with JS from the IdP and will use
that IFRAME to communicate with the IdP, authenticating
with Alice's identity (e.g., cookie). Similarly, Bob's
PeerConnection instantiates an IFRAME to talk to the IdP.
</t>
</section>
</section>
</back>
<!--
On Thu, Sep 22, 2011 at 5:45 AM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> Hi EKR,
>
> (As an individual)
>
> Thanks for posting the draft.
>
> I am missing a few security issues that I think should be considered.
>
> 1. The attempt to overload the links in an domain by concentrating
> traffic on the domain by choosing peer-pairs. Not that I think there is
> any real protection against this other than limit the flows to their
> "fair" share.
>
> 2. Configuring RTCP or other automatically sent traffic to high
> bit-rates. Especially under conditions where continued consent can't be
> determined.
>
> Cheers
>
> Magnus Westerlund
>
>
>
-->
<!-- drill down -->
</rfc>
<!-- Continuing consent per Magnus -->
<!-- TODO: Require key destruction -->
| PAFTECH AB 2003-2026 | 2026-04-23 13:15:36 |